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    Hello,<br>
    <br>
    not sure I understood your problem. The respective tutorial is for
    using freeswitch as a media server for kamailio. That means services
    like voicemail, conferencing, announcement, etc.<br>
    <br>
    Now, from your email I understand that you want all calls to go to
    freeswitch and it will do the routing, right?<br>
    <br>
    Cheers,<br>
    Daniel<br>
    <br>
    On 8/12/10 9:24 AM, Abhishek [ABK] Kumar wrote:
    <blockquote
      cite="mid:AANLkTinVEDYLpa6a_mjifoXvK5x=dXM=UJsjZyRdYcY0@mail.gmail.com"
      type="cite">Hi SiP-Users,<br>
      <br>
      I got kamailio installed and configured to work using mysql with
      FreeSWiTCH using this HowTo wiki<br>
      <a moz-do-not-send="true"
href="http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms">freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
        [Asipto - SIP and VoIP Knowledge Base Site]</a> : <a
        moz-do-not-send="true"
href="http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms">http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms</a><br>
      <br>
      This is also reference on FreeSWITCH portal and was the only
      decent tutorial I was able to find on integrating Kamailio with
      FreeSWITCH.<br>
      <br>
      Followed everything successfully, but I have a slight
      confusion/problem in its wonderful working...<br>
      <br>
      From all tests I performed, I perceived following activities:<br>
      a.) it handled the extensions added to it very well, and carried
      over the call to destination extension<br>
      b.) if destination extension is also added to it, call established
      without intervention of FreeSWITCH<br>
      c.) if extension is not added to it but FreeSWITCH, the call
      reached FreeSWITCH but was rejected there<br>
      d.) Red5Phone (developed over RED5 Server, @Google-Code) could
      dial with success but didn't received any<br>
      <br>
      Now,<br>
      <i><b>for point (a)</b></i> Thank You!<br>
      <i><b>for point (b)</b></i> it should be transferred to FreeSWITCH
      and that should be allowed to establish call, otherwise all
      dial-plans,etc. goes waste<br>
      <i><b>for point (c)</b></i> may be some settings in Wiki are
      Obsolete OR New, could you please point it out<br>
      <i><b>for point (d)</b></i> it may be due to point (b)<br
        clear="all">
      <br>
      -- <br>
      <br>
      --&nbsp;<br>
      Regards,<br>
      Abhishek Kumar<br>
      <a moz-do-not-send="true"
        href="https://sites.google.com/site/abhikumar163/"
        target="_blank">https://sites.google.com/site/abhikumar163/</a>
      <div>
        <br>
      </div>
      <div><br>
        --&nbsp;<br>
        --------------ABK-----mail.signature--------------------<br>
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          href="http://www.blogger.com/profile/06276198262605731980"
          target="_blank"><img moz-do-not-send="true"
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        <div>
          -----------------------------------------------------------<br>
          ~=ABK=~</div>
      </div>
      <br>
      <pre wrap="">
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</pre>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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